similar to: Asterisk and sip router integration

Hello! We're looking for a solution to reliably load balance our Asterisk boxes. So far we've been using a hodge-podge of directing different services to different boxes/IPs, but eventually I'd like to consolidate things so we can present a single IP address to the outside world. My question is - how do we go about doing that? I've read a lot of things like load-balancing via

Hi, I remember seeing a T38 Gateway application for Asterisk 1.6 floating around, but I can't seem to find it again. Does anyone have any pointers to it? I really want to be able to send an incoming T38 connection directly to the PSTN. Thanks. -- James

On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan <mjordan at digium.com> wrote: > On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna <jlamanna at gmail.com> wrote: > > Also, how big does the cache in frame.c grow to? > > I've recompiled with MALLOC_DEBUG on that server: > > > > asterisk -rx "memory show summary" > > > > .... > >

> > Its up to 5.8G of resident memory with 28321 calls processed. > The OOM killer is going to kill this soon at this rate (8GB RAM machine). > This seems like a pretty serious problem. > It looks like I'll need to restart asterisk every night.... Hi the number of cpu cores that you see with top times 512Mbyte is the level of ram that's needed e.g. a hp-gen8 with 2 octo

Hi, I'm coming up with ideas about building a cluster of asterisk servers, and am exploring the virtualization option. I'm curious to know some real-world data about how many extensions a VMWare install on good hardware could support. I've seen stories about how the hypervisor timeslicing can wreak havoc on call quality at some point. Is this really the case? If so, what's a

Hi, I've noticed that if I have a multi-line linksys (942 or 962) phone with the same sip registration mapped to each line key, that if all the lines are full the phone will accept another call. I would expect the phone to respond with "busy" so the call would to directly to voicemail. Has anyone else experienced this and know of a workaround? I know it seems like an

Also, how big does the cache in frame.c grow to? I've recompiled with MALLOC_DEBUG on that server: asterisk -rx "memory show summary" .... 1780466242 bytes (1780181594 cache) in 2352909 allocations in file frame.c ... Seems like a ridiculous cache. On Mon, Nov 24, 2014 at 9:02 AM, James Lamanna <jlamanna at gmail.com> wrote: > cat /proc/cpuinfo lists 4 cores. >

Hi, I'm having an issue where some phones behind a sonicwall are auto-congesting. The status on "sip show peer" shows ping times anywhere from 80ms all the way up to 1100ms. PCs behind the same firewall have a ping time of about 30ms to the PBX itself. Does anyone know if the sonicwall is inserting delay into the SIP signaling path and lagging the OPTIONS messages for qualify?

Hello, I'm currently testing a so-called VQ RTCP-XR feature from a a SIP hardphone. When a phone has enabled this feature, it would send a SIP PUBLISH to its SIP Server letting this server dispatch to whatever is needs to. These messages are sent during calls but may also be sent when a call is over. At the moment, I'm using Asterisk to serve these SIP phones so my Asterisk box

Hi, Does anyone have any good empirical data suggesting what the maximum number of PRI calls (incoming and outgoing) without hardware echo cancellation can be handled on a single box is? I have a TE410P T1 (1st gen) card and I'm seeing interesting errors of D-Channels going down and then coming back up (See below). I've looked at the number of simultaneous calls at each of these points,

Hi, I've been trying to fix NOTIFY replies (specifically keep-alives) in 1.4.42 (I can't upgrade to 1.8.x at the moment for various reasons). I've noticed for user agents that have a VIA header with a different port than the port the NOTIFY was sent from, the NOTIFY reply will always be sent back to that port, which is incorrect. (Sonicwalls and other routers love to do this, even

Hi all, Disregard my previous posts, I've consolidated everything here. I'm having terrible performance issues with samba over a WAN (point-to-point T1 link). Doing a copy of a 2MB file from a samba server to a linux client running smbclient takes over 5 minutes. SCPing the same file takes seconds. The server is running samba version 3.0.25c with kernel 2.6.16.18. I've put up a set

Hi. Between 2 linux (2.6.11 client and 2.6.14 server) machines connected by a 100Mb link I get samba performance copying a file from the client to the server through a smbmounted share of around 4.2MB/s Is this to be expected? Or can it be improved (and if so, how?) I've tried tweaking SO_(SND/RCV)BUF (after reading numerous articles on samba performance...), but it doesn't seem to have

2017-02-16 14:27 GMT+01:00 Joshua Colp <jcolp at digium.com>: > On Thu, Feb 16, 2017, at 09:11 AM, Olivier wrote: > > Hello, > > > > I'm currently testing a so-called VQ RTCP-XR feature from a a SIP > > hardphone. > > > > When a phone has enabled this feature, it would send a SIP PUBLISH to its > > SIP Server letting this server dispatch to

Hello, Our university has to upgrade soon its old Nortel PBX's which holds around 10,000 extensions tied to 5 PBXes. Up to now we thought about commercial solutions but now there is a window openning for open source solution. However, I need examples to convince that this solution is feasible, and preferably at other universities. Are there any pointers for such installations?

Hi, I have about 10 Cisco 7960s behind a PIX 506E (IOS v6.3) firewall. After some period of time, asterisk says that some of them are unreachable, and the phones lose their registration. The only way to make the phones recover is to clear the NAT translation tables for the phones on the PIX (clear xlate...) Does anyone know how to fix this? As you can imagine, it is quite annoying. And it does not

Hi all! I am looking for some software that would work as a proxy between a SIP device (SIP phones and ATA boxes) and the Asterisk system running IAX. The reason is that I can only get IAX trow the firewalls, and can't change the settings. One solution I am using are getting several Asterisk system communicate trow IAX bout in this case would I rater have every persons computer act as a proxy

Hi, I have a serious problem with Asterisk 1.4.18. Every so often, usually after Asterisk has been running for a few days consistently, phones start dropping registrations. However, when this happens, doing a "sip show peer" on those extensions shows them as "OK". Therefore, I have no way to tell this problem is happening until customers start calling. The only way to fix it is

> Date: Wed, 28 Jan 2009 13:11:19 -0600 > From: "Danny Nicholas" <danny at debsinc.com> > Subject: Re: [asterisk-users] SIP Registrations broken on 1.4.22.1? > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users at lists.digium.com> > Message-ID: <D32AD473FC574B41AE6A842E46549174 at db0005> >

I'm starting a new project similar to a previous project where I used OpenSER to front a bunch of Asterisk servers. Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely candidates. I'm leaning towards OpenSIPS because it's in EPEL so I can install it with yum. Also, because I think the name sounds more 'professional' when discussing architecture with clients :)